[..] |
ASTERISK-07000: [patch] func_realtime.c does not build with MTX_PROFILE |
ASTERISK-07001: [patch] aelparse will not build with MTX_PROFILE enabled |
ASTERISK-07002: [patch] chan_sip buffer overrun |
ASTERISK-07003: [patch] some little G.711 optimizations (idea by Rizzo) |
ASTERISK-07004: [patch] res_snmp.c format fix |
ASTERISK-07005: [patch] app_echo DTMF_BEGIN/END, HTML and IMAGE frames support |
ASTERISK-07006: [patch] fixed spelling |
ASTERISK-07007: [patch] deadlock between msglist_lock and loglock in logger.c |
ASTERISK-07008: [patch] [need disclaimer] Cannot set month range. |
ASTERISK-07009: After issuing asterisk -rx "logger reload" fast after another asterisk blocks |
ASTERISK-07010: [patch][post 1.4] cannot get peer by hostname from realtime storage |
ASTERISK-07011: deadlock on call park timeout |
ASTERISK-07012: app_chanspy causes deadlock after some time of active load |
ASTERISK-07013: Hangup timedout |
ASTERISK-07014: [patch] UniqueId for Leave event |
ASTERISK-07015: [patch] When SIP contains MIME header with Content-Type: multipart/mixed;boundary=..., call gets connected with no audio |
ASTERISK-07016: [patch] consistent MeetMeLeave Manager event |
ASTERISK-07017: Unnecessary playback of temporary greeting |
ASTERISK-07018: Asterisk crashes when jabber server shuts down. |
ASTERISK-07019: INVITE parsing fails when options exist in URI before @ sign. Fails carrier interop testing with Sonus GWs. |
ASTERISK-07020: WARNING[21992]: chan_zap.c:7503 zt_pri_error: PRI: XXX Missing handling for mandatory IE 20 (cs0, Call State) XXX |
ASTERISK-07021: can't get dtmf callerid |
ASTERISK-07022: Music on hold class not being set properly |
ASTERISK-07023: No Ringing Sound When Dialing a PRI Channel Over IAX |
ASTERISK-07024: [patch] setting the sourceaddress parameter in iax.conf doesnt function |
ASTERISK-07025: [patch] DB_DELETE dialplan function |
ASTERISK-07026: SIPCHANINFO(peername) Function returns null value |
ASTERISK-07027: [patch] new application: SayCurrency |
ASTERISK-07028: chan_sip.c rev 29904 does not compile. |
ASTERISK-07029: Call initiated by Callfile - Calleridnum changed by Dialplan - DISA resets to original |
ASTERISK-07030: Problem to send SMS from an internal S0-Bus. Channel doesn't hangup. |
ASTERISK-07031: fast hangup causes indefinite ast_channel_spy_trigger_wait |
ASTERISK-07032: chan_sip.c version 29904 breaks authentication |
ASTERISK-07033: Dial timers fail on SIP-SIP calls |
ASTERISK-07034: incorrect SIP dial string in extensions.conf |
ASTERISK-07035: Auth Fails on Failover |
ASTERISK-07036: Console '-x' flag loses data |
ASTERISK-07037: [patch][post 1.4][incomplete] Initial patch for doing simple message flagging |
ASTERISK-07038: [patch] More deprecated apps |
ASTERISK-07039: deadlock somewhere, channels get locked |
ASTERISK-07040: Auto fallthrough, channel 'SIP/230-6f32' status is 'UNKNOWN' |
ASTERISK-07041: "receive text <timeout>" timeouts first but then returns the right string |
ASTERISK-07042: meetme:admin_menu: option 2 and 3 are not working |
ASTERISK-07043: MixMonitor() related segfault on agent hangup |
ASTERISK-07044: [patch] Missing indications for countries available from libtonezone |
ASTERISK-07045: Crash when using GROUP application. |
ASTERISK-07046: [patch] print a human-friendly warning message about codecs |
ASTERISK-07047: reduce IAX_DEFAULT_REG_EXPIRE slightly to improve NAT/PAT reliability |
ASTERISK-07048: some time there is a .txt file and not the audio file |
ASTERISK-07049: [patch] Formatting/spacing for chan_zap.c |
ASTERISK-07050: [patch] Use ast_random in chan_jingle |
ASTERISK-07051: [patch] Remove unused var in res_jabber |
ASTERISK-07052: [patch] Fix utils/Makefile |
ASTERISK-07053: Possible callingpres logic problem |
ASTERISK-07054: [patch] pri_event_hangup structure fields not always initialized |
ASTERISK-07055: deadlock and, may be, memory leak |
ASTERISK-07056: This issue is relate do 0007084 MixMonitor patch with app_queue.c |
ASTERISK-07057: [patch] Say number needs correct grammar for British English |
ASTERISK-07058: [patch] SIP calls going through the same device can have duplicate channel names |
ASTERISK-07059: mysql & sip.conf &nat |
ASTERISK-07060: Voicemail deletion issue |
ASTERISK-07061: Hint function in Dialplan? Should it be in SIP.CONF? |
ASTERISK-07062: [patch] Race condition in app_meetme generates a SIGSEGV |
ASTERISK-07063: IAX fails behind nat and dynamic IP because the same port source is always used |
ASTERISK-07064: [patch] [post 1.4] Binding RTP on a public IP |
ASTERISK-07065: [branch] Asterisk will not compile on Linux (on s390/s390x) |
ASTERISK-07066: [patch][post 1.4] add handling "423 Interval Too Brief" to Asterisk |
ASTERISK-07067: Unable to use pattern matching with HINT function |
ASTERISK-07068: [patch][post 1.4] destroy channel on ZT_EVENT_REMOVED |
ASTERISK-07069: [patch] jitterbuffer support for "other" channels. |
ASTERISK-07070: [patch] global variables minor cleanup in chan_sip.c |
ASTERISK-07071: [patch] typo in comment utils.c (recursive instead of reentrant) |
ASTERISK-07072: [patch] Q921 debugging options |
ASTERISK-07073: chan_sip can't handle Cirpack KeepAlive Packets |
ASTERISK-07074: res_jabber cannot login on jabber server |
ASTERISK-07075: pciradio.c does not compile with linux 2.6 |
ASTERISK-07076: [patch] fxotune either doesn't recognize or doesn't display -l or -m options, causes subsequent tuning failure |
ASTERISK-07077: Segmentation fault on regular basis |
ASTERISK-07078: Since 1.2.8 I can't use reinvite with one of my SIP prividers |
ASTERISK-07079: SIP Channel Lock issue |
ASTERISK-07080: [patch] [need disclaimer] memory leakage in callerid.c |
ASTERISK-07081: [patch] Call received on a link declared in network mode : wrong call state after answering the call |
ASTERISK-07082: realtime mysql and cdr mysql in trunk are both broken |
ASTERISK-07083: [patch] add T309 to libpri : maintain calls in case of brief alarms |
ASTERISK-07084: asterisk does not obey libdir |
ASTERISK-07085: __load_resrouce kills startup. |
ASTERISK-07086: [patch] record spooling support in cdr_addon_mysql.c |
ASTERISK-07087: [patch] If you page many devices and the caller hangs up immediately, some devices stay stuck in the dynamic meetme |
ASTERISK-07088: Asterisk doesn't fork and become a daemon |
ASTERISK-07089: If CallerID name contains single quote Asterisk answers call but doesn't send it anywhere |
ASTERISK-07090: [patch] Double Ringing of Members |
ASTERISK-07091: [patch][post 1.4] app_queue: linear strategy |
ASTERISK-07092: If CallerID name contains single quote Asterisk answers call but doesn't send it anywhere |
ASTERISK-07093: app_dumpchan not functional |
ASTERISK-07094: make install target incompatible with packaging |
ASTERISK-07095: API Manager close connections in less than a minute with medium to high load. |
ASTERISK-07096: [patch] ANI not set with callerid= in some channel drivers |
ASTERISK-07097: [patch] handle ast_calloc failure |
ASTERISK-07098: Asterisk 1.2.8 will not compile on SuSE 10.0 |
ASTERISK-07099: [patch] ODBC VM deletes msg 0 when leaving message too short |
ASTERISK-07100: Agents logged in with AgentCallbackLogin hangs up when pressing * |
ASTERISK-07101: DTMF not passed properly when included in blindxfer featuremap |
ASTERISK-07102: [patch] asterisk should not fail on chan_zap config problems |
ASTERISK-07103: [patch] [post 1.4] No AMI event sent on dialplan variable changes through AGI |
ASTERISK-07104: socket_read() typos after search/replace. |
ASTERISK-07105: cannot compile with libc6-dev version 2.3.6-7 |
ASTERISK-07106: [patch] Asterisk sends INVITE instead of BYE if canreinvite=yes |
ASTERISK-07107: [patch] [post 1.4] app_meetme unique-id for relating calls to the same conference |
ASTERISK-07108: [patch] ast_channel_bridge in channel.c doesn't check AST_FEATURE_REDIRECT before deciding if a native bridge is suitable |
ASTERISK-07109: [patch] AGI command STREAM FILE does not log the file being played. |
ASTERISK-07110: qcall - pbx_spool.c always drops call when any sort of progress is received |
ASTERISK-07111: Can't match bye to request on a Native Bridge. |
ASTERISK-07112: [patch] janitor-style patch, use ast_strlen_zero |
ASTERISK-07113: Agent hung after getting call from Manager API Originate |
ASTERISK-07114: [patch] get gmake to re-read configuration |
ASTERISK-07115: [patch][post 1.4] cdr_csv log files splitting |
ASTERISK-07116: Call screening in app_dial does not delete the screening file when using option 'n' |
ASTERISK-07117: [patch] output of 'asterisk -rx "show xxxx"' gets chopped |
ASTERISK-07118: [need commit] additional sounds for Asterisk-Sounds-Extra |
ASTERISK-07119: Agent Logs to a non-existent extension |
ASTERISK-07120: Loading Zaptel modules breaks SIP completely |
ASTERISK-07121: [branch] bugs in AEL code |
ASTERISK-07122: [patch] duplicate lock in auth_fail(). chan_iax2.c |
ASTERISK-07123: [test_this_branch] rev. 33340 |
ASTERISK-07124: CDR vars are read-only unneccesarily. |
ASTERISK-07125: [patch] duplicate assignment in __get_from_jb(). chan_iax2.c |
ASTERISK-07126: [patch] Bulgarian indications |
ASTERISK-07127: [post 1.4] even though asterisk binds to 0.0.0.0 it wont allow IAX client registration to eth alias |
ASTERISK-07128: broken audio when spying any channel |
ASTERISK-07129: incoming calls on misdn channel are rejected |
ASTERISK-07130: [patch] lock.h __ast_cond_timedwait(). Incorrect handling of ETIMEDOUT |
ASTERISK-07131: Usage of NVLineDetect application |
ASTERISK-07132: Usage of NVLineDetect application |
ASTERISK-07133: [patch] CallerID name sent over PRI to Panasonic DBS locks channel |
ASTERISK-07134: meetme list returns user id's with a 0 prefix, but does not accept commands with 0 prefixes |
ASTERISK-07135: Missing newline on log message |
ASTERISK-07136: [patch] db1-ast makefile does not obey cflags |
ASTERISK-07137: [patch] bug in get_destination |
ASTERISK-07138: asterisk -rx "EXTENSIONS RELOAD" locks up |
ASTERISK-07139: [patch] recent changes to chan_sip, cause registrations to be broken |
ASTERISK-07140: [patch][post 1.4] Allow reversal of message presentation order, i.e. from newest first to oldest first and optional busy msg |
ASTERISK-07141: Meetme Patch to make Quiet option apply when users are kicked off the conference |
ASTERISK-07142: [patch] fix DEBUG_CHANNEL_LOCKS build (eliminate warnings) |
ASTERISK-07143: No ringing state notification |
ASTERISK-07144: Using ODBC voicemail storage and latest Centos 4.3 updates to mysql server, Asterisk crashed on startup. |
ASTERISK-07145: Malformed callerid string causes SIP clients to silently drop packets |
ASTERISK-07146: Attended transfer does not free up the agent in queue |
ASTERISK-07147: [patch] AGI scripts send bogus information to Asterisk |
ASTERISK-07148: [patch] [post-1.4] add manager command to sent a text message to a channel |
ASTERISK-07149: Asterisk crash a lot |
ASTERISK-07150: [patch] strnlen multiple definitions |
ASTERISK-07151: [copyright violation] broken audio when spying any channel |
ASTERISK-07152: [patch] res_odbc: remove unneeded error messages + some simplifications |
ASTERISK-07153: [patch] res_osp: remove unneeded error messages |
ASTERISK-07154: [patch] res_smdi: handle ast_calloc failure |
ASTERISK-07155: [patch] memset is not needed with ast_calloc |
ASTERISK-07156: Asterisk crashes every couple of days -- sip callback agents, iax callers |
ASTERISK-07157: [patch][post-1.4] IAX native bridge and NAT |
ASTERISK-07158: Asterisk crashes every couple of days -- sip callback agents, iax callers |
ASTERISK-07159: [patch] app_directory does not support ODBC voicemail storage |
ASTERISK-07160: [patch] ast_func_read segfault |
ASTERISK-07161: SIP CANCEL fails due to wrong Contact: URI |
ASTERISK-07162: asterisk crash with core dump |
ASTERISK-07163: [patch] Using italian language, voicemailmain suddenly exits when pressing a key on the phone |
ASTERISK-07164: [patch] Asterisk crashes on AddQueueMember with local channel |
ASTERISK-07165: [feature request] cdr_custom+mkfifo locks entire asterisk |
ASTERISK-07166: exiting on OS X produces poll errors |
ASTERISK-07167: [patch] pressing 0 while leaving a voice mail and waiting causes msgXXXX.txt file to be left behind. |
ASTERISK-07168: [patch] [post-1.4] Added Incremental Dialing to app_dial (Untested) |
ASTERISK-07169: [patch] External Password Notification (Instead of Replacing Default) |
ASTERISK-07170: [patch][post-1.4] External Password Notification (Instead of Replacing Default) |
ASTERISK-07171: [patch] Local channels get hung on call forwards |
ASTERISK-07172: [patch] [post-1.4] New "queue_log_name" parameter in logger.conf to set the name of the queue_log file |
ASTERISK-07173: Redirect other calls from dial-plan (without using manager interface) |
ASTERISK-07174: [patch][post-1.4] Short flash times - make it a friendly option in zconfig.h |
ASTERISK-07175: [patch] [post-1.4] "MacroExclusive" application - thread-safe Macro() |
ASTERISK-07176: Reload Clears Paused Agents |
ASTERISK-07177: [patch] [post-1.4] QueueLog app so you can write to the queue_log from your dialplan |
ASTERISK-07178: [patch] main Makefile - fix adsi target |
ASTERISK-07179: [patch] pressing DTMF features while prompts are playing cause havoc on the console |
ASTERISK-07180: Asterisk crash in parse_dial_string 9 times out of 10 |
ASTERISK-07181: [patch] wav49 format issue |
ASTERISK-07182: [patch][post-1.4] app_redirect |
ASTERISK-07183: Unable to unload and reload chan_agent from the cli |
ASTERISK-07184: MP3Player command has stopped functioning in new upgrade |
ASTERISK-07185: [patch] On Busy or Congestion after Answer we send wrong SIP messages - we act like the call isn't answered |
ASTERISK-07186: [patch] Update to the CDR RADIUS README file |
ASTERISK-07187: [patch] pointer problem in code2str() function |
ASTERISK-07188: [patch] Regression issue: ast_base64encode requires larger buffer than before |
ASTERISK-07189: Cannot put a call on hold and resume |
ASTERISK-07190: Running MythTV on the same system prevents chan_sip from operating. |
ASTERISK-07191: [patch] calc_metric and max_penalty |
ASTERISK-07192: [patch][post 1.4] escalate over penalty if no one answers |
ASTERISK-07193: MixMonitor displays the contents of a buffer that seem to have been freed already, or that was never initialized |
ASTERISK-07194: app_channelredirect causes crash |
ASTERISK-07195: SIP MWI indication; Message-Account field mangled |
ASTERISK-07196: [patch] Asterisk build errors and fixes for gcc 4.0.2 and Solaris 2.8 64bit |
ASTERISK-07197: [patch] A bug in Polish voice mail |
ASTERISK-07198: Voicemail box scanning causes significant and unnecessary i/o activity |
ASTERISK-07199: Sip peer identity hijack |
ASTERISK-07200: Response for call limit is incorrect |
ASTERISK-07201: [patch] Invalid handling of multiple secrets for peer/friend entries |
ASTERISK-07202: chan_h323 channel driver looks up SRV records for outbound calls |
ASTERISK-07203: [PATCH] externpass Doesn't Save User Password |
ASTERISK-07204: [patch] timing issue (race) with poke/pong for very close peers can cause peer to be declared unreachable |
ASTERISK-07205: CAN NOT Hangup channel in time. |
ASTERISK-07206: [patch] timezone reference in private.h is incorrect for Solaris build |
ASTERISK-07207: Bridged SIP RTP stream not redirected back to asterisk due to missing REINVITE message |
ASTERISK-07208: zaptel 1.2.6 compile error on linux PPC YDL |
ASTERISK-07209: [patch] Venezuelan indications and tone information |
ASTERISK-07210: [patch] allow SIP Spiral to work instead of causing a '482 Loop Detected' condition |
ASTERISK-07211: [patch] asterisk segfault after a signout/signin back |
ASTERISK-07212: [patch] [post 1.4] Add support for Visdn Drivers and better support for Euro ISDN |
ASTERISK-07213: Error in the math expressions with strings with non-standard characters |
ASTERISK-07214: [patch] Realtime queue members not added |
ASTERISK-07215: [patch] Provide support for /etc/{default,sysconfig}/zaptel file with module-dependant options for 2.6-based kernels |
ASTERISK-07216: [patch] Realtime Voicemail (Re)Connection |
ASTERISK-07217: MixMonitor causes segfault with voicemail |
ASTERISK-07218: no audio output when spying on bridged IAX2 channels with 'b' option |
ASTERISK-07219: can't parse WWW-Authenticate: Header correctly |
ASTERISK-07220: compilation breaks with DBUSYDETECT_MARTIN and DBUSYDETECT_TONEONLY enabled |
ASTERISK-07221: [patch] [post 1.4] Voicemail sound filename customization |
ASTERISK-07222: [patch] voicemail odbc storage retrieve_file problem |
ASTERISK-07223: Fast Dialing Feature |
ASTERISK-07224: Stripping digits does not work in NoOp |
ASTERISK-07225: crash on bridging channels, one of which does not have a jitterbuffer enabled |
ASTERISK-07226: [patch][post 1.4] addon for automon |
ASTERISK-07227: [patch] cannot transfer calls with r36148 |
ASTERISK-07228: Forbbiden |
ASTERISK-07229: func_odbc (backport) on 1.2.9.1 sometimes crashes Asterisk |
ASTERISK-07230: Current SVN does not compile and link in all of the needed files for pbx_dundi.so |
ASTERISK-07231: Verbose(${SIPPEER(status)) output is truncated |
ASTERISK-07232: ResetCDR(w) resets CDR data before writing it |
ASTERISK-07233: [patch] Build fails for zttranscode.c on CentOS 4.3 |
ASTERISK-07234: [patch] segfault due to not checking structure existance |
ASTERISK-07235: Meetme Crashing |
ASTERISK-07236: Bug 0006011 is not fixed in 1.2.9.1 |
ASTERISK-07237: [patch] See ${RDNIS} via DumpChan() |
ASTERISK-07238: [branch] Devicestate not working correctly |
ASTERISK-07239: vmexten is always set to "asterisk" |
ASTERISK-07240: [patch] Parked call fails to return to extension that parked it after a timeout |
ASTERISK-07241: [patch] menuselect changes |
ASTERISK-07242: [patch] A transfer SIP call present one way audio problem until hold and resume the call |
ASTERISK-07243: [patch] actually remove autoconfig.h on dist-clean |
ASTERISK-07244: wctdm24xxp is not included in sysconfig script |
ASTERISK-07245: [patch] Remove $(or) logic from Makefile, as it is only available in the very latest make command |
ASTERISK-07246: [patch][post 1.4] Move SIP registration information out of sippeers table into it's own |
ASTERISK-07247: NOTIFY sent to the wrong IP address/port |
ASTERISK-07248: recursive forward leads asterisk to lock channels |
ASTERISK-07249: BUG #0007435: Parked call fails to return to extension that parked it after a timeout |
ASTERISK-07250: Transfers from queue agent to another queue break app_queue |
ASTERISK-07251: Asterisk hangs up on parked calls when using applicationmap |
ASTERISK-07252: [patch] Formatting for a few log messages |
ASTERISK-07253: [patch] Parsing fails if From header contains angle brackets |
ASTERISK-07254: FreeTDS problem patch |
ASTERISK-07255: Asterisk Native Sounds ulaw play back with artifacts |
ASTERISK-07256: AMD leaves Iax channels open after check and never hang up |
ASTERISK-07257: [patch] AMI over HTTP fails with Internet Explorer |
ASTERISK-07258: CLI -> Sip show peer XXXXX |
ASTERISK-07259: CLI -> sip show inuse |
ASTERISK-07260: CLI -> sip show registry |
ASTERISK-07261: [patch] An Agent transfering call via SIP transfer causes a segfault or deadlock in queue/agent system |
ASTERISK-07262: chan_sip not closing channel when RTP goes idle in channel not exting in real life |
ASTERISK-07263: CPU Load Issues - Related to Auto-Attendant prompts? |
ASTERISK-07264: [branch] Reinvite does not manage RFC 2833 -> wrong payload |
ASTERISK-07265: app_amd broken in the case of no frames/digital silence |
ASTERISK-07266: [patch] sounds/Makefile hates solaris |
ASTERISK-07267: Asterisk crashed after OOH323 Call |
ASTERISK-07268: Use of FILTER crashes the server |
ASTERISK-07269: Usage of REGEX crashing Asterisk |
ASTERISK-07270: [patch] "fix" -addons 'install' make target |
ASTERISK-07271: [Patch] JabberStatus does not work if a resource is passed |
ASTERISK-07272: app_playback does not respect language setting |
ASTERISK-07273: ACK from asterisk still wrong |
ASTERISK-07274: [patch] Static Queue member 'pause' status is not stored with persistentmembers=yes |
ASTERISK-07275: [patch] AMI over HTTP returns malformed response header |
ASTERISK-07276: Asterisk fails to follow RFC3581 when SIP client sends ;rport in REGISTER message |
ASTERISK-07277: AEL2 not properly escaping commas |
ASTERISK-07278: [patch] Play announcement to caller just before bridging |
ASTERISK-07279: DESTDIR Support in Makefile |
ASTERISK-07280: No Audio from SIP to ooh323 direction |
ASTERISK-07281: RTP Packets go through Asterisk |
ASTERISK-07282: asterisk -rx reload hangs when console logging disabled |
ASTERISK-07283: Cisco 7912 phone continues ringing after sip call has finished |
ASTERISK-07284: Send CANCEL with wrong URI on ACK |
ASTERISK-07285: [patch] MySQL addons not building |
ASTERISK-07286: coredump on ast_context_remove_extension when compiled with MALLOC_DEBUG |
ASTERISK-07287: randomly crash ( core dumped ) when get over 15 callin from zaptel PRI |
ASTERISK-07288: One way audio, zap->*->IAXY Dropping incompatible voice frame |
ASTERISK-07289: [patch] typo in help dial |
ASTERISK-07290: Parked call timeout disconnects parked caller |
ASTERISK-07291: Output of 'sip debug' always goes to console |
ASTERISK-07292: VoicemailMain Application records too fast when in wav format. |
ASTERISK-07293: [branch] dropped just after INVITE call doesn't finishes correctly |
ASTERISK-07294: Spelling error in func_db.c |
ASTERISK-07295: [patch] Decoding / Encoding / sending AOC-D messages ("Advice of charge") |
ASTERISK-07296: [patch][post-1.4] AOC-D ("Advice of Charge - During call") passthrough |
ASTERISK-07297: Cannot build chan_zap with pri_cpe signalling |
ASTERISK-07298: core show commands don't work |
ASTERISK-07299: [branch] GROUP_COUNT returns invalid result on calls from queue to agentcallback or Local channel |
ASTERISK-07300: [patch] [post 1.4] Add CLID spoof option to Dial application |
ASTERISK-07301: [patch][post 1.4] add max_open_files=xxx to asterisk.conf |
ASTERISK-07302: Asterisk doesn`t compile |
ASTERISK-07303: Call drops on ACK-OK loop |
ASTERISK-07304: chan_h323 build broken by rev 47053 |
ASTERISK-07305: [patch] [post-1.4] MailboxExists() should be a dialplan function |
ASTERISK-07306: [patch] Typo in zapata.conf.sample |
ASTERISK-07307: [patch] VOICEMAIL_FILE_MODE not applied |
ASTERISK-07308: Introduce missing cause "Non-selected user clearing" |
ASTERISK-07309: [branch] New channel state transition to "Ringing" erroneously reported as Newchannel event |
ASTERISK-07310: [patch] Speed increase by doing a directory walk instead of repeated stat(2) calls |
ASTERISK-07311: safe_asterisk should stop if a configuration error occurs. |
ASTERISK-07312: automon appears to ignore MONITOR_EXEC outbound |
ASTERISK-07313: Zaptel Problem from UAE |
ASTERISK-07314: SIP 302 responses do not fill CALLERID(RDNIS) |
ASTERISK-07315: transmit_refer function uses uninitialized memory |
ASTERISK-07316: [patch] no DTMF detection on Zap outbound calls when called device does not send proceeding |
ASTERISK-07317: Buffer not always initialized |
ASTERISK-07318: [patch] ENUMLOOKUP function can't handle multiple records with the same order and priority |
ASTERISK-07319: [patch] Asterisk crashes when rtcp struct fails allocation due to too many files open |
ASTERISK-07320: Dial with option D does not appear to work in 1.2.9.1 |
ASTERISK-07321: [patch][post 1.4] Add read-only options and manager event |
ASTERISK-07322: [patch] Zaptel-1.0.10 no longer builds with kernel 2.6.15 |
ASTERISK-07323: app_mixmonitor.c:220: ast_channel_spy_read_frame:channel.c:3820: memcpy () |
ASTERISK-07324: [patch] make install fails with syntax error: unexpected end of file |
ASTERISK-07325: Segfault with Outbound Jingle Calls |
ASTERISK-07326: Regexp for sip addresses containing other characters than digits |
ASTERISK-07327: Music-on-hold from files switches files toward outside caller during SIP transfer-to-Park |
ASTERISK-07328: [patch] whitespace fixing |
ASTERISK-07329: [patch] ulimit CLI tool |
ASTERISK-07330: [patch] SSL connection for cdr_mysql |
ASTERISK-07331: sms receiving does not work |
ASTERISK-07332: [patch][post 1.4] NUMERAL function - numeral selection for Russian and Ukrainian language |
ASTERISK-07333: RTCP storm while trying out-of-the box echo test |
ASTERISK-07334: trunk unable to register with iax or sip |
ASTERISK-07335: bad logical operation when manipulating least significant word from ntp timestamp |
ASTERISK-07336: Asterisk dumped core on ast_db_put in db.c on no-load system |
ASTERISK-07337: [patch] call not terminated after timelimit expires |
ASTERISK-07338: [patch] Variable mis-name in enum.txt |
ASTERISK-07339: no manager hold / unhold event on SIP channel |
ASTERISK-07340: implement mipsel cross-compiling in the main Makefile |
ASTERISK-07341: Problems with parsing SIP URI Good ( 1.2.4 ) Bad ( 1.2.9.1 ) |
ASTERISK-07342: [patch] Unable to build Zaptel 1.2.7 due to massive number of errors |
ASTERISK-07343: Sending CANCEL instead of a BYE |
ASTERISK-07344: Forced bypass switch problem |
ASTERISK-07345: /dev/zap directory not being built. |
ASTERISK-07346: not sending progressing dss1 messages |
ASTERISK-07347: Agent can not log off and can not receive call when he leaves a meet me room |
ASTERISK-07348: Spaces in setvar= command not accepted |
ASTERISK-07349: TIMEOUT() values not respected inside Macro call from Dial() command |
ASTERISK-07350: Sound files minutes.gsm and seconds.gsm do not get installed |
ASTERISK-07351: rtptimeout setting fails to hang up a call dialed with AGI script |
ASTERISK-07352: Sip registration causes Asterisk to hang |
ASTERISK-07353: unable to bridge voip calls to PSTN using TDM4xxp |
ASTERISK-07354: Asterisk killed by SIGSEGV when accessing existing database record. |
ASTERISK-07355: [patch] The 'm' flag to Monitor does not have an effect when recording in ulaw |
ASTERISK-07356: q.931 FACILITY message with caller name in facility IE never sent from pri_net side |
ASTERISK-07357: SIP URI parsing patch to release branch "breaks" interoperability |
ASTERISK-07358: Call Waiting switching during monitoring locks out the channel |
ASTERISK-07359: MACRO changes UNIQUEID |
ASTERISK-07360: Callee's sip response TEMPORARILY_UNAVAILABLE (480) makes asterisk estabilish the connection and send congestion tone |
ASTERISK-07361: French hang up detection problem |
ASTERISK-07362: FreeBSD Asterisk-port crashes when transfering/forwarding to Local/ext |
ASTERISK-07363: E1 ISDN Won't Start - !! Got a UA, but i'm in state 1 |
ASTERISK-07364: [patch] Normalize fields separator in IEs dump functions - fix a trace |
ASTERISK-07365: Call drops when calling from E1 euroisdn to iax extension |
ASTERISK-07366: [patch] [post-1.4] Additional information for EXITWITHTIMEOUT and EXITWITHKEY events log |
ASTERISK-07367: [patch] Crash when an IAX2 call from another Asterisk server fails (due to missing application) |
ASTERISK-07368: [patch] [post-1.4] Allow an offset for SIP_HEADER |
ASTERISK-07369: logger.conf parse errors incorrectly reported |
ASTERISK-07370: park, parkedcall is causing asterisk to crash |
ASTERISK-07371: chan_zap.c is reporting PRI channels in use on ring event, no inbound calls until asterisk restart |
ASTERISK-07372: [patch][post 1.4] IAX2 Native Bridge Enhancements |
ASTERISK-07373: The ability for macro to accept an input |
ASTERISK-07374: Set the CID to a ZAP extension |
ASTERISK-07375: Asterisk will stop sending any packets for serveral seconds |
ASTERISK-07376: dialling more than two * in a phone number causes a seg fault |
ASTERISK-07377: Channel dosen't terminates after 487-Response from Grandstream GXP-2000 |
ASTERISK-07378: Problem with CALLERID handling in version > 1.2.0 |
ASTERISK-07379: app_queue unstable when retry=0 |
ASTERISK-07380: chan_sip goes boom |
ASTERISK-07381: [patch][post 1.4] Add SIP_HEADER_COUNT function and enhance SIP_HEADER to accept an index |
ASTERISK-07382: [patch] Possible SQL injection in addons/res_config_mysql.c |
ASTERISK-07383: Expand the status for Pause and Unpause |
ASTERISK-07384: zttranscode doesn't compile on linux 2.4 |
ASTERISK-07385: [patch] zaptel kernel module depends on 2.6 module functions |
ASTERISK-07386: Converted malloc calls to ast_malloc / ast_calloc in manager.c |
ASTERISK-07387: Asterisk crashes with high load when MixMonitor() is used |
ASTERISK-07388: Problem calling from an IAX client to a SIP client on 2 different *'s, connected by DUNDi |
ASTERISK-07389: SIP Phone not accepting 3/2 digit extension |
ASTERISK-07390: PrivacyManager app doesn't check the incoming CID matches the minimum length |
ASTERISK-07391: [patch] Behave rationally when substring length is negative |
ASTERISK-07392: Unmute through *1 no longer works |
ASTERISK-07393: [patch] Skinny hold not implemented |
ASTERISK-07394: [patch] 'b' option creates recordings and cdr entries for "ringing" part of the call, when it shouldn't |
ASTERISK-07395: When dialing from console, Asterisk doesn't put the contact as it has to |
ASTERISK-07396: Wrong Caller ID after Attended Transfer |
ASTERISK-07397: asterisk -rx "show queues" occasionally causes segfault |
ASTERISK-07398: Support for t38 only call |
ASTERISK-07399: [patch] Only check for kernel sources if you are building some modules |
ASTERISK-07400: [patch] coredump on hangup when compiled with MALLOC_DEBUG |
ASTERISK-07401: [patch][post 1.4] Monitor() only input or output stream |
ASTERISK-07402: Authentication in register request fails if it consists of multiple lines |
ASTERISK-07403: labels for goto not working in macros |
ASTERISK-07404: Asterisk cannot find the 'hint' priority when using realtime extensions |
ASTERISK-07405: [patch] Use INSTALL_PREFIX to install firmware |
ASTERISK-07406: 'musicclass' setting is ignored in realtime sip table |
ASTERISK-07407: [patch] [penging update to trunk] Finished Czech say and voicemail |
ASTERISK-07408: [patch] Fastart unexpectedly disabled : flag mask overlapping |
ASTERISK-07409: use of chan_agent in combination with transfers causes deadlock |
ASTERISK-07410: [branch] AEL2 parser doesn't see macros in .conf language files |
ASTERISK-07411: [branch] Standalone AEL2 compiler |
ASTERISK-07412: [patch] coredump on blind transfer unless compiled with DEBUG_CHANNEL_LOCKS |
ASTERISK-07413: Notifications sent with wrong Content-Type |
ASTERISK-07414: [patch] GET VARIABLE does not return global dialplan variables |
ASTERISK-07415: [patch][post 1.4] SIP support for PRI_CAUSE on Hangup() of unaswered calls... |
ASTERISK-07416: Asterisk crashes when attended transfer doesn't read data and call ended(autoservice related) |
ASTERISK-07417: [patch] Zaptel misreports channel in battery drop state as available, then refuses to open it |
ASTERISK-07418: Add little more infos for doxygen |
ASTERISK-07419: [patch][post 1.4] control both legs after a bridge |
ASTERISK-07420: [patch] duplicate queuelog |
ASTERISK-07421: Call failures with branch 1.2 r38420 chan_sip.c |
ASTERISK-07422: [patch] advanced_options() destroys msg_cfg used for context and cid info before it's used |
ASTERISK-07423: [patch][post-2.0] IAX2 remote variables |
ASTERISK-07424: [patch] hard-safe -> hard-unsafe lock order detected |
ASTERISK-07425: [patch] Assign default value to gMediaWaitForConnect |
ASTERISK-07426: REGEX argument parsing isn't working |
ASTERISK-07427: [patch] Debian does not have /var/lock/subsys |
ASTERISK-07428: [patch] Erroneous rename event when blind transfering to parkext |
ASTERISK-07429: output of ${SIPPEER(peer:status)) is trimmed |
ASTERISK-07430: Random 403 errors under load with Grandstream phones (see 0004343) |
ASTERISK-07431: Channel locks up when FastAGI does SET CONTEXT, SET EXTENSION, and SET PRIORITY |
ASTERISK-07432: Goto fails in applicationmap (res_features) |
ASTERISK-07433: Duplicate applicationmap assignment causes weird behaviour |
ASTERISK-07434: [patch] app_voicemail triggers error messages when trying to lock non-existing path |
ASTERISK-07435: waitexten can not work fine in agi |
ASTERISK-07436: Agents don't get localized voice prompts |
ASTERISK-07437: moh reload causes music to cut out and error to be produced |
ASTERISK-07438: For loop increment not working right |
ASTERISK-07439: Backport STRPTIME function to 1.2 |
ASTERISK-07440: Dialing certain numbers with 'r' set in the Dial command doesn't remove the ringtone when the other side answers. |
ASTERISK-07441: [branch] AEL reverse compiler Enhancement Request |
ASTERISK-07442: CDRs scrambled on attended transfers |
ASTERISK-07443: [patch] seg fault when doing transfer and there is no "referred by" header field from transferer |
ASTERISK-07444: app_directory does not work with odbcstorage |
ASTERISK-07445: MOH not starting sometimes in file mode |
ASTERISK-07446: no RTP audio during a call |
ASTERISK-07447: Improve strings reported by ast_state2str |
ASTERISK-07448: crash * on accepting call with H323 |
ASTERISK-07449: Asterisk crashing in meetme several times a day |
ASTERISK-07450: Bug in Dutch syntax! |
ASTERISK-07451: ACD login Logout not working using polycom soft button |
ASTERISK-07452: SIPPEER{xx:status} - space, Dial - feature req., __TRANSFER_CONTEXT - question |
ASTERISK-07453: [patch] Chan_sip doesn't send subscription-state in notify_with_sipfrag for all messages |
ASTERISK-07454: Segfault under moderate load on a 64-bit platform |
ASTERISK-07455: QueueStatus when using realtime |
ASTERISK-07456: When calling meetme Newcallerid is sent Before Newchannel |
ASTERISK-07457: [patch][post 1.4] ControlPlayback option to start at an offset and report the offset where playback stopped. |
ASTERISK-07458: [patch] incorrect path to zaptel includes |
ASTERISK-07459: [patch] fix signedness issues in libss7 |
ASTERISK-07460: [patch] app_dial doesn't create priv-callerintros directory needed for privacy mode |
ASTERISK-07461: using sendtext across 2 machines |
ASTERISK-07462: [patch] Cannot unset H245 tunneling flag |
ASTERISK-07463: Multiple line ringing has issues |
ASTERISK-07464: [patch][post 1.4] Language support for Brazilian Portuguese |
ASTERISK-07465: SIP channels accumulating |
ASTERISK-07466: Cannot hear music on hold from my agi script |
ASTERISK-07467: [PATCH] Add manager event JabberEvent when a message is sent or received |
ASTERISK-07468: [patch] sip notify to reboot grandstream |
ASTERISK-07469: using Dial(local/xxx/n) causes core dump with xfer |
ASTERISK-07470: Courtesytone plays to work side of call on parked call pickup |
ASTERISK-07471: [patch] setting usetls and usesasl to no result in a segfault |
ASTERISK-07472: Transfering on a local channel doesn't work properly |
ASTERISK-07473: Asterisk core dump using ast_aji_send |
ASTERISK-07474: [PATCH] Add a manager command so that you can send jabber messages |
ASTERISK-07475: using queues with local channels and MixMonitor causes segfault |
ASTERISK-07476: Variables not passed to AGI script on call forward |
ASTERISK-07477: callid[80] too small when using B2BUA |
ASTERISK-07478: Ghost blf inuse bug... An easy way to not get calls at work... |
ASTERISK-07479: [patch] COMPLETECALLER and COMPLETEAGENT don't have all the information described in doc/queuelog.txt |
ASTERISK-07480: Latest SVN (8/7/06) chan_jingle incoming call from google talk dumps core |
ASTERISK-07481: T.38 passthrough is not working between two Sipuras 2100 |
ASTERISK-07482: hex encoded uri not decoded |
ASTERISK-07483: codec_g729a.so coredump in SVN trunk |
ASTERISK-07484: incorrect 488 response contact header |
ASTERISK-07485: Add Credit for my work |
ASTERISK-07486: [patch] The "regseconds" field in the iax realtime db is not as useful as it could be. Here's a stab at improving its usage |
ASTERISK-07487: Frontrange's Goldmine 6.7 SIP Client |
ASTERISK-07488: [patch] no audio on calls from google talk to Asterisk |
ASTERISK-07489: [patch] fix a segfault when calling that from from an AGI |
ASTERISK-07490: With no one logged into the queue, there is no music on hold for the first caller to the queue. |
ASTERISK-07491: wct4xxp.o: unresolved symbol t4_tasklet |
ASTERISK-07492: After a ForkCDR asterisk does no longer get the correct values for CDR(userfield) and CDR(accountcode) |
ASTERISK-07493: Asterisk crashes on HangUp due to bad RAM |
ASTERISK-07494: astmanproxy and res_jabber have some conflict |
ASTERISK-07495: AMI returns AgentComplete and HangUp events when agent answers the call |
ASTERISK-07496: [patch] func_odbc does not return correct value at first request after ODBC connection has been down |
ASTERISK-07497: [patch] Cannot transfer via REFER |
ASTERISK-07498: CLID and SRC fields incorrect in CDR table when setting CallerID from dialplan |
ASTERISK-07499: [patch] Allow bigger names for music on hold classes |
ASTERISK-07500: Incorrect locking causes a race condition |
ASTERISK-07501: [patch][post 1.4] res_segfault - handle segfault with a command of choice |
ASTERISK-07502: [patch] [post-1.4] The frequently requested arguments to Gosub |
ASTERISK-07503: [patch] closing fd twice with FAGI in use under good load causes deadlock |
ASTERISK-07504: MixMonitor crash |
ASTERISK-07505: [patchj] unload sla* apps on unload app_meetme.so |
ASTERISK-07506: I mad changes in the channels SIP and OH323 |
ASTERISK-07507: [patch] Increasing the maximum number of persistent members per queue |
ASTERISK-07508: [patch][post 1.4] AMI action QueueStatus does not show the strategy used in the queue |
ASTERISK-07509: [branch] Make chan_h323 live! |
ASTERISK-07510: Redirecting Local channels to Meetme causes a crash when executing "core show channels" |
ASTERISK-07511: no audio after returning from a dial command executed though AGI |
ASTERISK-07512: [patch] properly using LDLIBS |
ASTERISK-07513: [patch] not stop asterisk load when not configuring |
ASTERISK-07514: [patch] not stop asterisk load when not configured |
ASTERISK-07515: [patch] res_config_pgsql.c small cleanup |
ASTERISK-07516: [patch] res_config_mysql make parse_config static |
ASTERISK-07517: [patch] format_mp3 install doesn't use $DESTDIR |
ASTERISK-07518: Alternative to naming new extensions when building a switch conditional |
ASTERISK-07519: Replace the Gosub command with an internal AEL2 command? |
ASTERISK-07520: DTMF not read correctly on Zap channels if busydetect=yes |
ASTERISK-07521: MixMonitor stops after attended call transfer |
ASTERISK-07522: Jitter buffer stopped on failed Native Bridge |
ASTERISK-07523: app_rpt.c : too few arguments to function ast_request |
ASTERISK-07524: Unable to build Zaptel svn version |
ASTERISK-07525: AEL2 can't parse hint with the and char (Zap/4&Zap5) |
ASTERISK-07526: ./configure --silent prints text |
ASTERISK-07527: 'make clean' rebuilds files |
ASTERISK-07528: Wrong parsing of SDP if multiple c= lines present |
ASTERISK-07529: IAX Channel audio stuck if realtime asterisk ge a database lock an wait for update or insert |
ASTERISK-07530: pri prefix settings seem to have no effect |
ASTERISK-07531: Codec Selection Issues Causes Garbled Audio |
ASTERISK-07532: Post hangup macro calls only execute partially if called from within another macro |
ASTERISK-07533: [patch] SMDI documentation clairfication |
ASTERISK-07534: 'make -C path/to/zaptel linux26' fails: incorrect PWD |
ASTERISK-07535: CALLERID on attended transfers |
ASTERISK-07536: can't compile trunk (2.4.21-27.0.2.EL kernel, redhat ES 3) with make v. < 3.81 |
ASTERISK-07537: [patch] some logging enhancements to app_queue |
ASTERISK-07538: chan_sip allows INVITEs from unknown peers |
ASTERISK-07539: [patch] 'w' is ignored in pulse dialing mode |
ASTERISK-07540: Consistent use of print_codec_to_cli |
ASTERISK-07541: Devicestate via manager interface incorrectly thinks an endpoint is on hold. |
ASTERISK-07542: Asterisk crashes when the odbc connection is lost and res_odbc or cdr_odbc is enabled |
ASTERISK-07543: Call limits |
ASTERISK-07544: No Newchannel event of caller channels |
ASTERISK-07545: Problems with receiving INVITE without SDP |
ASTERISK-07546: [patch] Applications/functions not picking up system timezone |
ASTERISK-07547: Coloring Ring Back Tone (early media) |
ASTERISK-07548: [patch] Update contrib/scripts/autosupport to use zttest -c |
ASTERISK-07549: [patch] reindent all zaptel code |
ASTERISK-07550: [patch] chan_zap plays dialtone even if channel seize fails |
ASTERISK-07551: [patch] workaround for race condition with disconnect supervision |
ASTERISK-07552: [patch] Suggested fix for formatting issue in make menuselect |
ASTERISK-07553: RFC2833 DTMF Events Out of Sequence = Duplicate Digits |
ASTERISK-07554: [bounty] RTP RFC2833 DTMF Events Out of Sequence = Duplicate Digits |
ASTERISK-07555: SIP -> SIP call hangs forever with "Blocking in: ast_waitfor_nandfds" |
ASTERISK-07556: chan_mgcp hangs for uknown reason |
ASTERISK-07557: SIP Interop with Sonus Gateways (npdi=yes causes improper SIP parse) |
ASTERISK-07558: [patch] Successful call park does not read back parking location |
ASTERISK-07559: function callback sender in advanced options (3) -> callback sender (2) |
ASTERISK-07560: [patch] Jingle channel dial attempt causes Asterisk to segmentation fault |
ASTERISK-07561: [patch] Sometimes missing events in queue_log |
ASTERISK-07562: Fixup minor issues in skinny show devices |
ASTERISK-07563: [patch] Allow a simple lookup for hint extensions |
ASTERISK-07564: Requests handling do not comply to RFC 2616 |
ASTERISK-07565: [patch] Channel flag ZT_FLAG_OPEN setting and clearing are not atomic |
ASTERISK-07566: chan_skinny interger overflow (dos/possible remote compromise) |
ASTERISK-07567: [patch][post 1.4] add Request-URI (RURI) handling to SIPCHANINFO() function |
ASTERISK-07568: make install should remove the modules prior to installing the new ones instead of complaining about the old stray ones |
ASTERISK-07569: [patch] show translation text fix |
ASTERISK-07570: Three way conferencing on asterisk |
ASTERISK-07571: [patch] Asterisk crashes with chan_skinny |
ASTERISK-07572: segfault in chan_iax2 |
ASTERISK-07573: Crash due to running out of stack space -- nested macros |
ASTERISK-07574: sip.conf port option is not honored |
ASTERISK-07575: [patch][post 1.4] Implementation of QSIG ETS 300 258 - ISO Path Replacement (ANF-PR) |
ASTERISK-07576: unload app_random.so segfaults * |
ASTERISK-07577: Asterisk 1.2.10 Crash - the MixMonitor issue |
ASTERISK-07578: Segfault-maybe due to voicemail |
ASTERISK-07579: [patch] Enable multiple file streaming when calling Read from the dialplan |
ASTERISK-07580: Attended SIP Transfer Call Teardown Issue |
ASTERISK-07581: call recording for agents when using AgentLogin is not consistent |
ASTERISK-07582: Modules cannot be loaded in SVN HEAD version |
ASTERISK-07583: [branch] Cisco 7920 Phone Screen not Cleared on Call Complete |
ASTERISK-07584: [patch][post 1.4] make optional to strip the (inter)national prefixes when in "dynamic" (local)dialplan mode |
ASTERISK-07585: problem with authorisation when reload asterisk which register => user:secret... to another asterisk |
ASTERISK-07586: PlaybackStatus() does not set $PLAYBACKSTATUS |
ASTERISK-07587: [patch] res_sqlite module for Asterisk - SQLite 2 support |
ASTERISK-07588: res_agi changes in 1.2.11 side effect in agi get variables |
ASTERISK-07589: Agent announcements with AnnounceOverride truncated. |
ASTERISK-07590: Call progress monitoring on outbound calls not detecting SIT tones |
ASTERISK-07591: Calls from Asterisk to some H.323 ATAs ring but don't complete |
ASTERISK-07592: call barge in |
ASTERISK-07593: additional skip time |
ASTERISK-07594: Inconsistant Manager API events |
ASTERISK-07595: Directory App Speak Extension |
ASTERISK-07596: Incomming calls generate two call proceeding messages |
ASTERISK-07597: asterisk does not recognize correct peer trunk |
ASTERISK-07598: [patch] Bugfixes for handling of channel variables |
ASTERISK-07599: [patch] Pickup not working properly after upgrade to 1.2.11, always assuming SLINEAR |
ASTERISK-07600: Bug Collection for AEL |
ASTERISK-07601: Function Regex returns NULL on negative match |
ASTERISK-07602: [patch] [post-1.4] Write to queue_log via AMI |
ASTERISK-07603: [patch] chan_alsa indent |
ASTERISK-07604: [patch] janitor: chan_oss indent |
ASTERISK-07605: [patch] update asterisk-addons for new loader |
ASTERISK-07606: [patch] fix the say number in Hebrew to play 200 as it should |
ASTERISK-07607: [patch] security fix for format string issue in app_record |
ASTERISK-07608: [patch][post 1.4] IAX2 is missing SRV support for outgoing calls |
ASTERISK-07609: [patch] "save dialplan" doesn't save labels |
ASTERISK-07610: [patch][post 1.4] super quiet option to meetme |
ASTERISK-07611: Wrong ROSE_DIVERTING_LEG_INFORMATION2 in libpri? |
ASTERISK-07612: [branch] Change of channel state takes several minutes |
ASTERISK-07613: Problems with issue 5577 |
ASTERISK-07614: [patch] Issuing a "reload" after making changes in queues.conf causes corruption and/or crash |
ASTERISK-07615: Operators ! and =~ do not work in 1.2, contrary to what README.variables says |
ASTERISK-07616: Extensions.conf parser ignores labels |
ASTERISK-07617: [patch] ast_pbx_outgoing_exten in pbx.c doesn't put any information about which call failed in CDR or channel variables |
ASTERISK-07618: Lots of _udp_. dns requests |
ASTERISK-07619: [patch][post 1.4] app_lookuplist - Match a number from the specified database family |
ASTERISK-07620: [patch] When forwarding voicemail, app_voicemail ignores the delete option. |
ASTERISK-07621: music on hold app apparenlty caches the music when it comes from a stream |
ASTERISK-07622: [patch] Optionally return values returned by query as dialplan vars |
ASTERISK-07623: [patch] Chinese rendition of day-of-month in ast_say_date_with_format_tw is wrong |
ASTERISK-07624: [patch] segfault when processing SUBSCRIBE MWI |
ASTERISK-07625: [patch] fix crash when mixmonitor tries to stop monitors after channel.c does the same on hangup |
ASTERISK-07626: Recordings terminated with DTMF are over-trimmed |
ASTERISK-07627: Attempt to set a read-only variable error message should say which variable. |
ASTERISK-07628: CDR(billsec), CDR(calldate), CDR(duration), CDR(end) are not available in dialplan or AGI |
ASTERISK-07629: PeerStatus Event unreliable |
ASTERISK-07630: ${AGENTBYCALLERID_{CALLERID(num)}} |
ASTERISK-07631: [patch][post 1.4] Patch to make Background() return a status code in the same way Playback() does. |
ASTERISK-07632: [patch] make inbound channel 'Ringing' state consistent with other channels |
ASTERISK-07633: [patch] MySQL error report when queries fail |
ASTERISK-07634: [branch] XCON+BFCP - Centralized Conferencing Specs and Binary Floor Control Protocol |
ASTERISK-07635: [patch][post 1.4] new ast_do_masquerade manager event |
ASTERISK-07636: CURL() does not timeout |
ASTERISK-07637: Issue 0005641 |
ASTERISK-07638: [patch] t.38 passthrough not working when endpoints are behind a NAT |
ASTERISK-07639: asterisk crash - related to MixMonitor usage |
ASTERISK-07640: asterisk crash - unkown reason |
ASTERISK-07641: Mixmonitor crashes as soon as it is invoked |
ASTERISK-07642: Core dump while loading dundi at startup and receiving manager event |
ASTERISK-07643: [patch] Correct Snom check-cfg notify message in sip_notify.conf.sample |
ASTERISK-07644: [patch][post-1.4] New AccountActive() function |
ASTERISK-07645: [patch][post 1.4] Configurable lock types for ast_lock_path - allows voicemail storage on SMB/CIFS mounts |
ASTERISK-07646: Meetme gives extra announcements |
ASTERISK-07647: [patch] dialling out to a cirpack gateway ends up in a crash |
ASTERISK-07648: [patch] manager sends Newchannel event instead of Newstate |
ASTERISK-07649: [patch] reload_config always returns 0 |
ASTERISK-07650: [patch] Chan_ooh323 needs to be taught about VLDTMF to compile against current trunk |
ASTERISK-07651: [patch] ast_waitforexten_full() have missed break within case |
ASTERISK-07652: [patch] Revision 41283 breaks _extension_match_core() on exact matches |
ASTERISK-07653: [patch] Invalid handle of zero value of file descriptor |
ASTERISK-07654: Function REGEX fails to match beginning of string |
ASTERISK-07655: SIP_HEADER crashes asterisk |
ASTERISK-07656: [patch] Fix race condion crash with get_member_status |
ASTERISK-07657: cli error -- No such command 'exit' |
ASTERISK-07658: [Patch] Packetization for chan_skinny |
ASTERISK-07659: [PATCH] Inherit channel variables during call forwards when app_queue goes through chan_local. |
ASTERISK-07660: Keypad does not become DTMF |
ASTERISK-07661: 302 Redirect from Transfer command passes diaplan flow on a timeout but continues to resend 302 redirect |
ASTERISK-07662: Used B channels restart failure after E1 span comes back up |
ASTERISK-07663: crash in chanspy |
ASTERISK-07664: Automatically set externip when asterisk is SIP client behind NAT router |
ASTERISK-07665: Using 'n' option in Queue() causes static members list to be called twice |
ASTERISK-07666: [patch] chan_iax2 ignores "sourceaddress" when "bindaddr" is set to INADDR_ANY (the default). |
ASTERISK-07667: [patch] res_musiconhold causes asterisk to hang in Solaris when compiled with zaptel drivers |
ASTERISK-07668: [patch] Compilation failures when building on Solaris with Zaptel drivers |
ASTERISK-07669: [patch] reload does not initialize configuration values |
ASTERISK-07670: chan_misdn + AGI + ChanIsAvailable(): "Too many open files" and open pipes |
ASTERISK-07671: Redirecting Number (RDNIS) problem |
ASTERISK-07672: Strong Noises during call transfer |
ASTERISK-07673: [patch] trunkgroup minor type error |
ASTERISK-07674: Add priority to Dial() |
ASTERISK-07675: Set(TIMEOUT(absolute) = seconds) doesn't seem to work while AbsoluteTimeout does |
ASTERISK-07676: pciradio.c include moduleparam.h missing |
ASTERISK-07677: segfault when zap channels are full (calls are Originate'd via AMI and exacerbated by app_amd) |
ASTERISK-07678: FUNC_ODBC returns mixed up query results |
ASTERISK-07679: channel.c set_format() exits prematurely under some circumstances. |
ASTERISK-07680: [patch] Adding dial arguments via channel variable DUNDIDIALARGS |
ASTERISK-07681: Asterisk crashing when using custom application when JB is enabled |
ASTERISK-07682: SIP transfer by an Agent fails and causes a segfault if show channels is issued |
ASTERISK-07683: [patch][post 1.4] Add conference start and conference end manager events |
ASTERISK-07684: SIPAddHeader sets last used header only |
ASTERISK-07685: transfering call originating from agent channel fails |
ASTERISK-07686: [patch] Unnecessary Warning 'Strange... The other side of the bridge is not a SIP channel' |
ASTERISK-07687: [patch] zttool show incorrect timeslot's numbers and fail when not all timeslots selected |
ASTERISK-07688: [Patch] enhancement to app_read, to still set the variable if the caller hangs up after entering digits |
ASTERISK-07689: [post 1.4][patch] Allow the Read() application to accept multiple prompts (seperated by an ampersand) |
ASTERISK-07690: [patch] advanced_options frees memory then uses and modifies it later in the routine. |
ASTERISK-07691: Unable to get Sip peers registered from ARA |
ASTERISK-07692: asterisk is crashing if spying of an active channels is stopped |
ASTERISK-07693: called number presentation has changed to 4 digits since 1.2.11 |
ASTERISK-07694: RTCP trying to remove non-existent schedule when its transmission failed |
ASTERISK-07695: [patch] Transfer capability is inherited by a channel after being transfered via atxfer |
ASTERISK-07696: [patch] Jabber timeout enhancements |
ASTERISK-07697: [patch] fix infinite loop in app_record when using '%d' option |
ASTERISK-07698: Beeing authenticated by false/no password |
ASTERISK-07699: [patch][post 1.4] Keep queue call statistics on reload |
ASTERISK-07700: SIP or IAX redirect from AGI script to meetme room results in no audio |
ASTERISK-07701: voicemail application fails to state the extension number in the chosen language |
ASTERISK-07702: [patch] Add MySQL safe flag to the CDR() function |
ASTERISK-07703: INSTALL_PREFIX needs fixed in Makefile |
ASTERISK-07704: GSM codec: formats/msgsm.h: xmc[48] not initialized?! |
ASTERISK-07705: two small tweaks for FreeBSD 6.1-STABLE |
ASTERISK-07706: portability fixes: time_t vs. long |
ASTERISK-07707: [patch] redhat/asterisk.spec is stale |
ASTERISK-07708: [patch][post 1.4] asterisk-sounds needs RPM spec file |
ASTERISK-07709: when make pulse dial before last digit there is a pause that not present in a dialed number. |
ASTERISK-07710: ast crashes/gets confused, under diff situations when using cisco phones. |
ASTERISK-07711: unchecked reference to linux/soundcard.h in muted.c |
ASTERISK-07712: [patch][post 1.4] CLI and Manager function to clear Queues statistics |
ASTERISK-07713: Can't set CDR userfiled |
ASTERISK-07714: [patch] turn off silence suppression if no channel owner |
ASTERISK-07715: [patch] [post-1.4] security enchancments when using exec, system, rename etc. |
ASTERISK-07716: Using the # to terminate recording disconnects you |
ASTERISK-07717: Trivial spelling error in voicemail.conf.sample |
ASTERISK-07718: [patch] both 404 and 503 code responses sent when extension not found |
ASTERISK-07719: Spelling errors in .conf files |
ASTERISK-07720: Spelling errors in docs |
ASTERISK-07721: Call disconnect from Queue causes Seg Fault |
ASTERISK-07722: [post 1.4] Return multiple records (rows) when executing ODBC queries |
ASTERISK-07723: RFC-compliant enumlookup macro is borked. |
ASTERISK-07724: CALLERID component input not validated |
ASTERISK-07725: It is impossible for a multi-codec phone to talk to a multi-codec phone although both phone have a common codec. |
ASTERISK-07726: [PATCH] ast_log() and ast_verbose() new-line mess |
ASTERISK-07727: [patch] asterisk crash randomly |
ASTERISK-07728: [patch] Asterisk native sounds distort consistently(ulaw) |
ASTERISK-07729: Asterisk crashes with IAX2 |
ASTERISK-07730: [patch] Redirecting a call with the management API can cause a call to hangup |
ASTERISK-07731: [patch] ast_io_remove() does not properly decrement number of fds |
ASTERISK-07732: [patch] replaced hard-coded path in contrib/init.d/rc.redhat.asterisk with variable |
ASTERISK-07733: [patch] chan_sip could stop responding to clients after 'sip reload' from CLI |
ASTERISK-07734: [patch] Using trunk and chan_agent causes crash upon hangup |
ASTERISK-07735: [patch] add jitterbuffer parameters to "sip show settings" output |
ASTERISK-07736: [patch] Use of PQescapeString() compromises thread safety |
ASTERISK-07737: record should return recorded length in seconds in a variable |
ASTERISK-07738: can't compile misdn drivers ( b410p ) |
ASTERISK-07739: MS Live Messenger Channel |
ASTERISK-07740: ast_config that holds variables from voicemail txt file is destroyed too early in advanced_options() |
ASTERISK-07741: [patch] pri show span command references invalid argv |
ASTERISK-07742: [patch] Far end BYE causes weird Reinvite and zombie channel |
ASTERISK-07743: [patch] zonedata.c does not contain data for Russian Federation |
ASTERISK-07744: "sip prune realtime peer 1234" and "sip show peer 1234" load doesn't work as expected. |
ASTERISK-07745: [patch] use a better comparison to see if db connection is up in res_config_pgsql.c |
ASTERISK-07746: dnsmgr.conf not detecting and updating address changes |
ASTERISK-07747: static char *math_descrip does not say, that app_math is deprecated |
ASTERISK-07748: BUG reporting system leaks of capability to report bugs in func_math |
ASTERISK-07749: [patch] [post-1.4] add new function POW() |
ASTERISK-07750: Chanspy hangup causes crash |
ASTERISK-07751: [patch] Voicemail message callback option loses last 3 digits of CID number |
ASTERISK-07752: Wrong CallerID passed to SIP phone |
ASTERISK-07753: NFAS primary D-channel always being maked as ACTIVE on startup |
ASTERISK-07754: [PATCH] the wrong context is searched for the 'o' and 'a' extensions when a user presses '0' or '*' |
ASTERISK-07755: [patch] Caller ID blocking fails: RPID settings from extension not carried to trunk |
ASTERISK-07756: app_page calls hangup after ~3 seconds |
ASTERISK-07757: Syntax for 'w' optional delay is not as described in meetme description |
ASTERISK-07758: [patch] Add New Zealand number-unobtainable indication |
ASTERISK-07759: [patch] vfork() in asterisk.c |
ASTERISK-07760: exit() after fork() |
ASTERISK-07761: Asterisk crashes in handle_request_refer() when an agent does a transfer |
ASTERISK-07762: TIMEOUT() is not working |
ASTERISK-07763: [patch] meetme uses inconsistent names for callerid information for manager events |
ASTERISK-07764: Asterisk segfaults on Agent hangup |
ASTERISK-07765: Names for callerid information reported by the manager are inconsistent |
ASTERISK-07766: activating applicatoion feature cause 100% system load |
ASTERISK-07767: [branch][post 1.4] introduce jittertargetextra option to jitter buffer via iax.conf |
ASTERISK-07768: [patch] $(realpath) requires make 3.81 |
ASTERISK-07769: Segfault when using an argument for the LIMIT value |
ASTERISK-07770: blindxfer => ## is not working (broken by r38687) |
ASTERISK-07771: [patch] Constification of the return value of a config function which ought not be modified |
ASTERISK-07772: Add option to not mark new messages as old after listening. |
ASTERISK-07773: warning messages in ooh323 test |
ASTERISK-07774: hangup cause 38 for source releases calls |
ASTERISK-07775: ooh323 does not work in failover test case if the first destination is an empty/no route device |
ASTERISK-07776: cancellation does not stop ooh323 dialing an empty/no route device |
ASTERISK-07777: Add rtp-packetization.txt to doc directory |
ASTERISK-07778: codec capabilities are not checked on call legs when making an Attended xfer (REFER) |
ASTERISK-07779: [patch] missing locking in zt_unregister |
ASTERISK-07780: [patch] fixed two OSP application bugs |
ASTERISK-07781: Make Realtime SIP Clusterable - Message Waiting Not Working |
ASTERISK-07782: [patch] app_meetme causes channel to hangup on last marked user exit |
ASTERISK-07783: Phone is ringing but dont have a conversation! |
ASTERISK-07784: Unable to reconnect to IAX when remote server was down fo sometime |
ASTERISK-07785: Asterisk crashes on handling many subscriptions at the same time |
ASTERISK-07786: "make menuselect" fails on OS X |
ASTERISK-07787: Chanspy making asterisk exit |
ASTERISK-07788: [patch] chan_alsa indent and update for new loader |
ASTERISK-07789: [patch[ janitor: chan_mgcp |
ASTERISK-07790: [patch] chan_oss update for new loader |
ASTERISK-07791: [patch] early return AST_MODULE_LOAD_DECLINE if config not exists |
ASTERISK-07792: [patch] return AST_MODULE_LOAD_DECLINE if config not exists |
ASTERISK-07793: [patch] format_mp3 fix building |
ASTERISK-07794: [patch] chan_misdn not return AST_MODULE_LOAD_DECLINE when misdn not configured |
ASTERISK-07795: 1.4.0b2 crashes on 'reload' if chan_skinny is loaded |
ASTERISK-07796: [patch] Voicemails forwarded via email to a mailbox with delete=yes are not deleted |
ASTERISK-07797: "Extras" sound files mis-encoded |
ASTERISK-07798: [patch] chan_sip unable to terminate call from servers using multipart/mixed Content-Types |
ASTERISK-07799: asterisk 1.4b2 not compiling (using make < 3.81 and configure script not catching it) |
ASTERISK-07800: Asterisk 1.4.0-beta2 - MySQL not working with an underbar or dash in a field name |
ASTERISK-07801: Crash in memory allocation when using a sufficiently old glibc (2.3.2) |
ASTERISK-07802: ulaw always the first codec in SDP, regardless of position in the peer's allowed codec list |
ASTERISK-07803: [patch] Update the maximum packetization values in frame.c |
ASTERISK-07804: [patch] Using ODBC storage, the message is not correctly entered into database if reviewed by sender |
ASTERISK-07805: Asterisk crahes and exits when attempting to extract IAX information from a SIP call. |
ASTERISK-07806: multiple incoming call to one dynamic queue member |
ASTERISK-07807: chan_spy with whisper causes strange behaviour requiring reboot |
ASTERISK-07808: [patch] Add option to logger to rename log files with timestamp |
ASTERISK-07809: MOH dir needs to be changed in default musiconhold.conf |
ASTERISK-07810: Command 'iax2 show provisioning' already registered |
ASTERISK-07811: app_queue keeps crashing |
ASTERISK-07812: Crash due to recording |
ASTERISK-07813: call and pickup groups should support 64 groups, but it doesn't. |
ASTERISK-07814: IAX2 peer qualify requires in-process reload to start polling peers. |
ASTERISK-07815: Zap channels locked when Agents are involved |
ASTERISK-07816: Asterisk 1.2.2 + asterisk-messaging-0.0.3_mod_x-files.tgz problem |
ASTERISK-07817: Asterisk startup hangs when registering first function |
ASTERISK-07818: [patch] addition to support timeout and warning into MeetMe |
ASTERISK-07819: Fresh install on Debian won't launch - .ael file problems |
ASTERISK-07820: memory leak in 1.2.12.1 after upgrading from 1.2.10 |
ASTERISK-07821: [patch] fixes for IMAP storage support |
ASTERISK-07822: Local queue members tagged as "invalid" and cannot be used |
ASTERISK-07823: [Patch] Add QueueSummary command to Queue manager interface |
ASTERISK-07824: Calls CANCEL'ed or BYE'ed before 180+183 result in 487 retries and timeouts |
ASTERISK-07825: Ability to set hints for non-device related extensions |
ASTERISK-07826: Remove dead code in mm_login |
ASTERISK-07827: DTMF fails via SIP for Meetme |
ASTERISK-07828: [patch] app_queue compares devices incorrect (dial string vs. no dial string) |
ASTERISK-07829: Asterisk crashes when i call a gtalk contact or get called by one. |
ASTERISK-07830: Autopause not working for queue members |
ASTERISK-07831: dsp.c listening for wrong fax frequency ...or.... what does faxdetect=outbound do? |
ASTERISK-07832: DTMF fails via SIP for Meetme and VoicemailMain (works via TDM04B card) |
ASTERISK-07833: OpenH323 hard codes signallingChannelCallTimeout |
ASTERISK-07834: [patch] Missing MemberName if setinterfacevar is set |
ASTERISK-07835: [patch] persistentmembers=yes breaks with membername |
ASTERISK-07836: [patch] Make chan_local report INUSE and NOTINUSE instead of UNKNOWN |
ASTERISK-07837: an-error-has-occured in extra-sounds are cut off at beginning |
ASTERISK-07838: [patch] the call is hangup instead of playing the warning sound |
ASTERISK-07839: Presence does not support RFC 4480 |
ASTERISK-07840: app_queue.so module must be explicitely unloaded then loaded again in the CLI to work properly with Local members |
ASTERISK-07841: Asterisk crashes when an H323 call (chan_h323) is queued and dispatched to a Local queue member. |
ASTERISK-07842: [patch] when doing blind transfer (REFER), TRANSFER_CONTEXT is ignored |
ASTERISK-07843: Core Dump in ? on Hangup After Transfer |
ASTERISK-07844: [patch] cross compilation on mips |
ASTERISK-07845: Core dump on hangup in socket_process after multi pbx transfer |
ASTERISK-07846: GoSub doesn't pass arguments |
ASTERISK-07847: Real ISDN Switching |
ASTERISK-07848: something in the new compile process is causing AST_DATA_DIR to be empty |
ASTERISK-07849: Call drops on transfer when coming from queue/local channels (broken by r31520) |
ASTERISK-07850: eventmemberstatusoff is included in the sample queues.conf but is invalid |
ASTERISK-07851: [patch] hanguponpolarityswitch hangs up on incoming call during ring phase |
ASTERISK-07852: [patch] crash when calling skinny phone, that is unregistered |
ASTERISK-07853: Asterik 1.2 - > asterisk 1.4 (trunk) ooh323 crash |
ASTERISK-07854: PRI Channels become unavailable if too many call files are queued |
ASTERISK-07855: [patch] Remove IS_NULL_STRING from res_monitor.c |
ASTERISK-07856: asterisk locks, taking most of the box down, when performing attendant transfers |
ASTERISK-07857: request disabling srv look up feature for oh323 |
ASTERISK-07858: Sip Peers and IAX Peers via mxml produce invalid xml |
ASTERISK-07859: [patch] Add option_debug checks before blindly writing to DEBUG logging channel |
ASTERISK-07860: [patch] hardwired dchan channel number |
ASTERISK-07861: T38 relay doesn't work between Audiocodes Tulip AC494 ATAs |
ASTERISK-07862: Chanspy/Linked List Crash or Deadlock |
ASTERISK-07863: Asterisk 1.2.12 multihomed one way sound |
ASTERISK-07864: [patch] Make a test call from the CLI, have a report written in the CLI and log directory |
ASTERISK-07865: [patch] Fix indenting to make code more clean |
ASTERISK-07866: [patch] possible double free on app_voicemail.c |
ASTERISK-07867: Asterisk crash and no core debug provided |
ASTERISK-07868: [patch] Asterisk deadlocks when channels starting an AGI script get hung up |
ASTERISK-07869: Crash in mutex lock on 64-bit platform |
ASTERISK-07870: Agent logoff not working |
ASTERISK-07871: [patch] The record# parameter in ENUMLOOKUP is ignored |
ASTERISK-07872: Core Dump Loading AEL2 Config |
ASTERISK-07873: asterisk doesn't register anymore, sip show registry status show "Request Sent" |
ASTERISK-07874: Delay after 5 second in conversation time |
ASTERISK-07875: Problem with forcing SIP_CODEC on an iax2 trunks |
ASTERISK-07876: Can't use channel variables in EXEC |
ASTERISK-07877: CANCEL is not forwarded to the outbound call leg |
ASTERISK-07878: [Patch] Add rollback Option to makefile |
ASTERISK-07879: Calls stalls hangup til absolote timeout is reached |
ASTERISK-07880: [patch]MeetMeAdmin description of flags does not match their function |
ASTERISK-07881: AEL2 - causes segfault when an exten defined in extensions.ael2 is followed by an empty block |
ASTERISK-07882: calls from Zap to IAX channel with T or t Dial flag locks channels if # pressed |
ASTERISK-07883: Remote-Party-ID has wrong screen tag values |
ASTERISK-07884: SPA942 g726 call problems |
ASTERISK-07885: Call to chan_local target crashes asterisk |
ASTERISK-07886: Segmentation fault using chan_visdn |
ASTERISK-07887: app_directed-pickup cannot pickup groupcalls |
ASTERISK-07888: RFC2833 DTMF no longer detected with seqno flips to 0 |
ASTERISK-07889: MOH /var/lib/asterisk/moh/ does not exist in any format |
ASTERISK-07890: Asterisk Server Crash 5 times in one day |
ASTERISK-07891: Random crashes on call |
ASTERISK-07892: No way to remove iaxy.bin from tarball |
ASTERISK-07893: [patch] Getting machine readable voicemail.conf information (email, extension) through manager interface |
ASTERISK-07894: [patch] say_alpha does not break when DTMF digit is keyed |
ASTERISK-07895: CLI verbosity level can not be decreased |
ASTERISK-07896: After ~20 - 30 minutes of load asterisk IAX deadlocked, and do not accept any more calls and not stop on "stop now". Needs kill |
ASTERISK-07897: RFC2833 DTMF Events not detected when [END] packet arrives first |
ASTERISK-07898: IF causes seg fault if label doesn't exist |
ASTERISK-07899: [patch] Extension for Realtime API |
ASTERISK-07900: [branch] RT Storage for voicemail - initial release |
ASTERISK-07901: set(TIMEOUT(absolute)) is not honored after a call to Dial() |
ASTERISK-07902: Check the status of the SIP devices |
ASTERISK-07903: [patch] minor documentation update for func_timeout |
ASTERISK-07904: Notice: sched.c:296 |
ASTERISK-07905: http://www.archivum.info/asterisk-users@lists.digium.com/2006-01/msg00671.html |
ASTERISK-07906: No Dialtone |
ASTERISK-07907: [patch] G.711 codec woes |
ASTERISK-07908: chan_sip.c causes asterisk to crash randomly in __sip_destroy when ast_string_field_free_all is called |
ASTERISK-07909: Switch fallthrough fails to function with patterns |
ASTERISK-07910: No bridge call |
ASTERISK-07911: ACK's transmitted by a SIP channel that is not outgoing can have To and From headers switched. |
ASTERISK-07912: Music is unclear and choppy |
ASTERISK-07913: rtpholdtimeout and rtcp packets |
ASTERISK-07914: Device get registered only if debug is activated for his IP address |
ASTERISK-07915: raddiag.tar.gz corrupted? |
ASTERISK-07916: Linking fails for muted.o on Mac OS X |
ASTERISK-07917: [patch] app_chanspy volume adjust broken for certain channel types |
ASTERISK-07918: [patch] Allow exiting from ChanSpy to a valid sigle digit extension (like meetme) and add a readonly mode |
ASTERISK-07919: Asterisk crash when receving or sending out call for Google Talk |
ASTERISK-07920: Asterisk won't build with DEBUG_CHANNEL_LOCKS |
ASTERISK-07921: Pickup feature not usable... |
ASTERISK-07922: [PATCH] w(timeout) parameter of app_meetme does not work as advertised |
ASTERISK-07923: dnsmgr stops refreshing after a reload |
ASTERISK-07924: Module embedding fails on Mac |
ASTERISK-07925: 'say load new' on console and absence of say.conf crashes asterisk |
ASTERISK-07926: Problems with silent-suppression (VAD) |
ASTERISK-07927: Asterisk crashes with jitterbuffer + mixmonitor |
ASTERISK-07928: channel hangs on FastAGI in Asterisk 1.4 |
ASTERISK-07929: RTP listener not getting cleaned up upon call teardown |
ASTERISK-07930: [patch] Crash with chan_gtalk during Echotest application |
ASTERISK-07931: new standard extension 'b' to execute at the beginning of a context |
ASTERISK-07932: Page() command kills dynamic MeetMe conference in 5 seconds due to (5) flag ... |
ASTERISK-07933: P2P connection is established when endpoints are using incompatible codecs |
ASTERISK-07934: CRASH - sched.c: Attempted to delete nonexistent schedule entry |
ASTERISK-07935: Asterisk, configured as ooh323 GW fails to register with Nortel CS2000 GK |
ASTERISK-07936: Segmentation fault when calling Dial with a Jingle channel |
ASTERISK-07937: Check return codes everywhere |
ASTERISK-07938: Out of idle IAX2 threads for I/O |
ASTERISK-07939: Critical - No audio issue with re-invite (wrong media address) |
ASTERISK-07940: [patch] Improve const-correctness |
ASTERISK-07941: [patch] Enable specifying SDP RTP IP address |
ASTERISK-07942: Asterisk crashes on app_queue.c |
ASTERISK-07943: Unable to have played sound in background (async with script) manner |
ASTERISK-07944: Asterisk Crashes with Glibc error on CentOS 4.4 |
ASTERISK-07945: [patch] Added bindaddr in configuraton setting in gtalk.conf |
ASTERISK-07946: [patch] ExternalIVR does not hangup the channel when asked to |
ASTERISK-07947: Crash in res_jabber |
ASTERISK-07948: [patch] trivial fix for a compilation problem in manager.c |
ASTERISK-07949: Frequent asterisk lockups |
ASTERISK-07950: Segmentation fault with mysql realtime |
ASTERISK-07951: SIP don't show real ip into cli |
ASTERISK-07952: asterisk stops transfering calls asterisk.c-2414 |
ASTERISK-07953: Hangup status not propagated to master of Local channel |
ASTERISK-07954: [feature request] No Uniqueid generated on Async Originate AMI request |
ASTERISK-07955: Sent wrong event "newchannel" in case of "newstate" |
ASTERISK-07956: Asterisk doesn't have any way to tell the dialplan what version it is |
ASTERISK-07957: [patch] ExternalIVR references non existant 'silence-10' sound file |
ASTERISK-07958: [patch] ExternalIVR does not respect channel language selection |
ASTERISK-07959: When playing periodic-announce in queue, agents phone stop ringing until the playback end |
ASTERISK-07960: TDM400P 4FXO. Trouble with incoming calls |
ASTERISK-07961: Check build options for Pthread usage |
ASTERISK-07962: Page funtion stops after a couple of seconds (MeetMe) |
ASTERISK-07963: Module unload causes segfault |
ASTERISK-07964: [patch] missing unregister for application |
ASTERISK-07965: SIP call termination fails due to From/To field mixup when callee hangs up |
ASTERISK-07966: [patch] Bogus ast_mutex_unlock |
ASTERISK-07967: User-Agent header with version |
ASTERISK-07968: [patch] log jitter buffer stats regularly for post-mortems, route qualification etc etc |
ASTERISK-07969: [patch] Jitterbuffer PLC fix for IAX2 channel and other issues with jitterbuffer |
ASTERISK-07970: codec g729 - unsupported format (when call TO skinny phone) |
ASTERISK-07971: DTMF digits are not recognized |
ASTERISK-07972: SDP information incorrect if canreinvite is globally undefined |
ASTERISK-07973: [patch] Asterisk to Gtalk audio shuts off after 30 seconds into call |
ASTERISK-07974: [patch] res_monitor does not unregister two manager commands |
ASTERISK-07975: ast_bridge_call not found on MacOSX |
ASTERISK-07976: Asterisk doesn't like CNAMEs or DNS servers that are down |
ASTERISK-07977: Space as the first character of the username part of a uri - valid? |
ASTERISK-07978: After a correct blind/attended transfer audio comes and go leaving audio holes. |
ASTERISK-07979: Improper charset in man pages |
ASTERISK-07980: Fail of ioctl |
ASTERISK-07981: Asterisk can't dial or register sip devices after a while |
ASTERISK-07982: Asterisk can't dial or register sip devices after a while |
ASTERISK-07983: Asterisk can't dial or register sip devices after a while |
ASTERISK-07984: [patch] Jabber crashes asterisk when getting a message with no "from" |
ASTERISK-07985: Page only forwards audio for 5 seconds |
ASTERISK-07986: parse_uri can crash asterisk |
ASTERISK-07987: WaitExten broken for sub-1-second times |
ASTERISK-07988: codec g_729a not backward-compatible to older glibc-versions |
ASTERISK-07989: [patch] outside user can't get auth for subscribtion when adding asterisk's account to their IM |
ASTERISK-07990: Selection of mutex type |
ASTERISK-07991: Check handling of strncpy calls |
ASTERISK-07992: [patch] extend ControlPlayback to fwd/rew file with different speeds |
ASTERISK-07993: 1.4 trunk with-odbc fails on RHEL4 |
ASTERISK-07994: Calling from SIP to ISDN via ZAP - hangup problem |
ASTERISK-07995: [patch] run macro when queue member connected |
ASTERISK-07996: Asterisk Crashes when meetme receieves ZAP Hangup Request |
ASTERISK-07997: Called side still ringing when calling side hangup |
ASTERISK-07998: asterisk 1.2.13 crash in calling a member |
ASTERISK-07999: SER adds port to call ID - still the same SIP dialog? |